So I received a message in my forum inbox asking about how to use a focusrite interface as a mixer to sum two mics when doing a profile. I thought maybe other people might benefit from the answers, so I thought I'd open a thread on it rather than answer just one person 
Also, I'm hoping @JackFocusrite might chip in, as he seems to be a good guy and very helpful here
The message linked to one of my posts in which I said that I'd circumvent the daw for the summing part, and simply use the focusrite unit to sum (link here: Amp Profiling Questions?). After looking at it again, I'm kind of reconsidering this.
I do believe that you should shift the mics around to get the best possible sound before doing anything else. Get them as much in phase as possible - or rather, get the best sound you can get using the tool of mic placement. This includes placement, blend, distance, possibly baffling, absorption, getting the cab up off the flor (or placing absortive material at the first reflection points etc etc, which I will not get into here in any detail.
Once you get to that point though, I see no problems in sweetening stuff a bit. I realized that this is also where there is a difference between making a profile, and recording to separate tracks, which you can blend, shift and EQ to sweeten the sound to a degree you might not be able to without those tools at your disposal.
Now, EQ (for a DSP-enabled interface) and blend can be done entirely in the focusrite mixcontrol software, in which you need to setup the correct routing. What I'd do is:
- Make a submix (click one of the unused "Mix [x]" tabs on the top to bring that up)
- Solo the two input channels in use, turn everything else down on that page (apart from the "Mix [x]" fader on the far right)
- Blend the two channels to taste
- EQ to taste, if you have a DSP interface (you would have to be input 1 and 2, I believe, in order to apply the built-in EQ effects. Those can still be set to accept line level inputs in the mix control software, useful if you use external preamps). And I think that in the "mix[x]" tab, I think you have to select "FX (anlg 1)" and "FX (anlg 2)" to get the EQ part.
- Open the "router" (there's a button on my interface in the far left, around middle of the window). Find an unused line output, find the corresponding button on the router (there are buttons for selecting what goes out of each output, like this: [button] -> Line output [x]). Click the button, select "Mix [x]".
- Plug that corresponding output into the Kemper return (or whatever it is we usually plug mics into when profiling) - you might need a TRS->XLR cable for this.
I THINK this should do it - but I've never tried it myself, and I'm going by my limited understanding here - @JackFocusrite, I'm hoping for confirmation or correction here
This method, however, is only really useful if you have a separate room for your loud amp, if you want to be able to hear what the mics are actually picking up. This is doubly true for any interface-DSP-EQ. The blend created so far (including any DSP-EQ) will NOT be picked up when recording - you have to do further wizardry in order to bake in the blend/EQ etc when recording (something about loopback, I think). I won't get into that, as I'm on thin ice as it is.
What I'd actually propose for most people is kind of (partly) the opposite of what I advised originally in the post linked above.
Since we're baking the blend etc into a single profile, I would actually profile through the DAW.
Now, this doesn't preclude you from doing the whole dance of proper mic placement, at all!! Get a GOOD sound upfront, before any wizardry. No, get a GREAT sound upfront.
For most of us, this means a lot of trial and error by way of RECORDING the mics, then adjusting placements. Lather, rinse, repeat. Take frequent breaks, of you will chase your own tail.
Look at some of the ideas in the third paragraph of this already way too long post.
Get your mic blend on your interface, not in the DAW. Don't get lazy already! Lock down the variables where they are intended to be locked as you go along!
It's a bit of back and forth. Back that SM57 off the grill a bit, the proximity effect is killing the sound. Now the SM57 is a bit lower in the blend withthe other mic, so adjust the gain on that input. As needed. Who knows, maybe it sounds even better without touching the knob.
Now, we have as good a starting point as we can get (and let's just assume that this work done of the front end should be about 95% of the time spent on the whole process - don't kid yourself).
So let's see if we can get it even better.
Record a snippet. IN THE STYLE YOU'RE AIMING AT WITH THE AMP SETTINGS YOU CHOSE INITIALLY. Don't go all Slash Solo on the death metal RHYTHM guitar / country chicken picking profile you initially wanted to create. And play WELL, darnit! Don't just noodle some half-arsed riff out because you want to get on with things. Play tight, play right. This goes for the whole process!
(man, I have WAAAY too little experience with "real world recording" to merit the semi-arrogant tone I write this in, haha :-))
Now you have a basis for doing some final sweetening.
IN THE DAW, on the recorded tracks you start to adjust stuff.
Maybe start out by delaying the close mic'ed track a bit, to eliminate the latency between the tracks caused by the different speaker-to-mic distances of the individual mics. Find the best delay setting, likely to get the attacks as much in time as you can between the two mics, eliminate comb filtering etc. Also: Insert a gain plugin on the master channel and adjust it to compensate for any changes in volume introduced by the delay plugin. Adjust several times to get it right. Always compare at the SAME level, or you will just pick the loudest one. Playback the result for a time, adjusting to the overall sound and tone. STOP PLAYBACK. turn off both the delay plugin and the gain plugin. Playback. Which one sounds better? Repeat the process if needed. Oh, and use a CLEAN delay with ONE repeat. No distortion, no tape effects, no hipass/lopass stuff. I'm talking about DELAYING the signal, not adding an "echo" effect. For instance, Logic Pro X has a "sample delay" which is the correct one to use here. Other daws should have the same stuff.
You might find that shifting the time around does nothing goof for the tone. Good. One less thing to overthink. One experience richer. Might work another time.
You also might find that now the blend is slightly off. Adjust that in the DAW if needed. WRITE A NOTE OF THE FADER POSITIONS.
Next, work the EQ of the tracks. There's a reason you used two mics, right? Not just because it was cool, right? (now I'm just downright insulting ;-)). You took mic A in position X because you like the midrange bite and mic B in position Y because you like the bottom end - or WHATEVER. That should give you at least some kind of pointer of your INITIAL idea of the role of each mic, which you can use as a starting point. So maybe you like the overall sound, but it's a bit bassy, and a bit lacking in brilliance. Listen to each mic. Which one has the BEST SOUNDING bass/lower mids? Which one has the BEST SOUNDING high mids/highs?
Whip out the EQ if it isn't perfect. Adjust each track as needed (but with both tracks playing), KEEPING IN MIND that you should compare at equal loudness between un-EQd and EQd signals.
BTW: A backwards trick I got from Kendal Osborne from the recording lounge podcast might be helpful with regards to EQ'ing while the two tracks are playing simultaneously - keeping in mind your vision of what role each mic was supposed to play:
- Too much bass? Adjust LEVEL until you get the right amount of mids and treble, then EQ down the bass back to where it sounds nice.
- Not enough bass? Adjust LEVEL until you get the right amount of bass, then EQ down the treble/mids to where it sound nice.
- vice versa if too much/not enough treble - whatever.
You might want to get more drastic and use the bass mainly from one mic, and mainly the mids/treble from another, or whatever.
I can't stress this enough: when comparing to the "original" recorded track, always compare at equal overall level. a gain plugin on the master is handy here. Find the correct gain setting, rest your ears for a bit, then compare.
Man, I'm getting way off track. I want to say too many things in one go. Sorry for that.
ANYWAY, when you've got a great sound from any (potential) EQ'ing, post-recording level adjustments and delay, then route the output FROM THE DAW to the output on the interface you want to plug into the Kemper.
Turn on the DAWs "monitoring" feature on the two tracks.
Profile.
Cross your fingers.
Test the profile.
Hopefully it will sound as intended.
I got so sidetracked that I'm sure I've missed something - if anybody is still with me, please point out all the mistakes, misinformation and general stupidity in this post.
